As is known, the public switched telephone network (PSTN) carries voice and/or video between origination points and destination points, for example, voice between origination and destination telephones. The PSTN is also adapted to carry dual-tone multi-frequency (DTMF) communications, generated, for example, by keypad pushes on a telephone or by a computer modem. The PSTN has historically included only analog telephone signals, either singly or as analog time-division multiplexed (TDM) telephone signals. The TDM analog signals can be multiplexed in a variety of ways, for example, as TDM analog signals corresponding to twenty-four analog telephone calls on a T1 telephone line.
The PSTN is a connection system, meaning that connections are formed and reserved between the origination and destination points covering an entire “session.” Even during periods of silence, the origination and destination points remain connected, resulting in relatively inefficient use of telephone wires and equipment.
Digital networks, for example the Internet, are connectionless systems, meaning that a connection is not established or reserved between an origination and a destination point at the time of the communication session. Instead, digital-network packets, which are sent from an origination point to a destination point, each travel to the destination point along a path determined by routers along the path. A path along which one digital-network packet is directed can be different than the path along which another digital-network packet is directed, even in the same communication. In this way, physical connections within a digital network are not reserved by either origination or destination points, but instead can be used by all origination and destination points. During idle periods between any origination point and corresponding destination point, the physical connections in the network are used by other origination and destination points, resulting in a more efficient use of physical equipment.
Noting this advantage, telephone companies have introduced digital networks within portions of the PSTN. As is known, some types of ‘gateways’ can translate from analog telephone signals or analog TDM telephone signals to digital-network packets and/or from digital-network packets to analog telephone signals or analog TDM telephone signals. Therefore, a telephone company can use a digital network within portions of the PSTN to improve equipment utilization. However, most conventional telephone equipment at the customer premises, i.e., origination and destination points, is adapted to send and receive only analog telephone signals.
Some forms of Customer Premises Equipment (CPE) are adapted to directly send and receive digital-network packets. For example, Internet telephones are able to communicate with Voice over Internet Protocol (VoIP) using a digital network. Some organizations thus distribute telephone calls with VoIP over a digital network within the organization, and convert the VoIP signals to analog telephone signals with a gateway when coupling the telephone calls to the PSTN.
A variety of telephone company equipment and CPE have one or more digital-network packet outputs for IP communications. Examples of such CPE include, but are not limited to, conference bridges, Internet telephones, and private branch exchange (PBX) systems.
It should be recognized that voice, DTMF, other audio signals, or video signals sent as digital-network packets need not be perfectly represented, i.e., need not have perfect data integrity, as is otherwise necessary with digital data. The voice, DTMF, other audio, or video signals, however, need to have a reasonable fidelity in order to be intelligible at the destination point.
As is known, digital-network packets progressing along paths as described above, can be lost or damaged. For example, a digital-network packet may be sent to a damaged router. Digital-network packets can also be re-transmitted, for example, by a router. Upon re-transmitting, the digital-network packet can be altered or damaged.
Therefore, it would be desirable to test audio and/or video signals within digital-network packets and equipment, which transports the digital-network packets, in the context of voice, DTMF, other audio, or video communications, which do not necessarily require perfect data integrity.
In order to test voice communications and/or voice communication equipment, which supports the IP protocol, conventional test systems acquire test voice clips, or, more generally, test audio clips. The test systems can acquire the test audio clips and compare them to expected audio clips. To this end the conventional test system acquires digital-network packets associated with a test audio clip, disassembles the digital-network packets to obtain real time protocol (RTP) payload portions of the digital-network packets having a test audio portion of the test audio clip, and re-assembles the RTP payloads to re-generate the test audio clip. In order to compare the test audio clip to an expected audio clip, the conventional test system converts the resulting re-generated test audio clip to the frequency domain with a Fourier transformation, and spectrally compares the test audio clip with the expected audio clip.
It will be understood by one of ordinary skill in the art that the disassembling and re-assembling of network packets as described above, as well as frequency domain processing, tends to require complex electronic circuits and software and tends to slow the testing throughput of digital-network packets.